• Journal of the Korea Institute of Information and Communication Engineering
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• v.18 no.12
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• pp.2871-2877
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• 2014

Real-time communication and diversification of multimedia application services using high-capacity data over the Internet demand high speed connection, and real-time and high-speed access internet service such as multimedia make Internet traffic increase rapidly. Meanwhile, QoS (Quality of Service) of internet service is not satisfied. In this situation, ISP (Internet Service Provider) has been required to improve service quality and extend network according to the user's needs. This request includes issue of extending network focused on increasing router and the number of routing table as well as simply extending bandwidth.

• The KIPS Transactions:PartC
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• v.13C no.2 s.105
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• pp.203-210
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• 2006

Fair packet scheduling algorithms supporting quality-of-services of real-time multimedia applications can be classified into the following two design schemes in terms of the reference time used in calculating the timestamp of arriving packet: Finish-time Design (FD) and Start-time Design (SD) schemes. Since the former can adjust the latency of a flow with raising the flow's reserved rate, it has been applied to a router for the guaranteed service of the IETF (Internet Engineering Task Force) IntServ model. However, the FD scheme may incur severe bandwidth loss for traffic flows requiring low-rate but strong delay bound such as internet phone. In order to verify the usefulness of the SD scheme based router for the IETF guaranteed service, this paper analyzes and compares two design schemes in terms of bandwidth and payload utilizations. It is analytically proved that the SD scheme is better bandwidth utilization than the FD one, and the simulation result shows that the SD scheme gives better payload utilization by up to 20%.

• The Journal of Korean Institute of Communications and Information Sciences
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• v.29 no.7B
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• pp.613-627
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• 2004

In this paper, we propose an Aggregate Fairness Maker (ARM) required for an Edge router to improve fairness of throughput among the flows of Assured Service in DiffServ with different round trip time (RTT) and we propose a user flow Three Color Marker (uf-TCM) as a flow marker that marks packets from the flow as green, yellow, or red. A yellow packet is the packet that consumes loss token in uf-TCM as well as that is demoted green packet in AM due to disobey the aggregate traffic profile. The proposed AFH promotes yellow packet to green packet or demotes green packet to yellow packet through the fair method without per-flow management, and it improves the feirness of throughput among the flows as well as link utilization. A yellow packet and a red packet have the same drop precedence at Core Router in our scheme. So we can use the RIO buffer management scheme. We evaluated the performance of our proposed AFM and the REDP Marker that was proposed to improve fairness without per-flow management. Simulation results show that, compared with the REDP marker, proposed AFM can improve performance of throughput fairness among the flows with different RTT and link utilization under the over-provisioning, exact-provisioning, and under-provisioning network environments at Multiple DiffServ domains as well as at Single DiffServ domain.

• Journal of KIISE:Information Networking
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• v.28 no.4
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• pp.540-549
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• 2001

Internet telephony service is very cheap and very easy to introduce the value-added service than the POTS, but is difficult to the QoS of telephone service. The existing Internet typically offers 'best effort' services only, which do not make any commitment about delay, packet loss and jitter. This paper compensates the low quality of the speech for packet loss or delay using SOLA algorithm in Internet phone. SOLA algorithm is a popular technique for Time Scale Modification of speech and audio signal. In the proposed algorithm, the receiver expands the received packet under resonable threshold, and hence compensates the QoS of speech. From the simulation, this algorithm can conceals packet loss considerably, and is also improved the quality of the speech.

• Journal of Internet of Things and Convergence
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• v.1 no.1
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• pp.1-4
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• 2015

This paper aims to present the mathematical model to find the mean web object transfer latency in the slow-start phase of TCP congestion control mechanism, which is one of the main control techniques of Internet. Mean latency is an important service quality measure of end-user in the network. The application area of the proposed latency model is the narrowband environment including multi-hop wireless network and Internet of Things(IoT), where packet loss occurs in the slow-start phase only due to small window. The model finds the latency considering initial window size and the packet loss rate. Our model shows that for a given packet loss rate, round trip time and initial window size mainly affect the mean web object transfer latency. The proposed model can be applied to estimate the mean response time that end user requires in the IoT service applications.

• Journal of the Institute of Convergence Signal Processing
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• v.10 no.3
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• pp.181-185
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• 2009

Recently, much attention to a self-organizing mobile ad-hoc network is escalating along with progressive deployment of wireless networks in our everyday life. Being readily deployable, the MANET (mobile ad hoc network) can find its applications to emergency medical service, customized calling service, group-based communications, and military purposes. In this paper we investigate a time slot exchange problem found in the time slot based MAC, that is designed for IEEE 802.11b interfaces composing a MANET. The paper provides a method to maintain the quality of voice call by providing a new time slot when the channel assigned for that time slot gets noisy with interferences induced from other nodes, which belong to the same and/or other subgroups. In order to assess the performance of the proposed algorithm, a set of simulations using the OPNET modeler has been performed assuming that the IEEE 802.11b interfaces are operating under a modified MAC, which is a time slot based reservation MAC implemented in the PCF part of the superframe. In a real-time voice call service over a MANET of a size 500 ${\times}$ 500 meter squares with the number of nodes up to 100, the simulation results are collected and analyzed with respect to the packet loss rate and packet delay. The results show us that the proposed time slot exchange protocol improves the quality of voice call over that of plain DCF.

• Proceedings of the KIEE Conference
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• 2008.11a
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• pp.379-381
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• 2008

TETRA(Terrestrial Trunked Radio) is a digital trunked radio standard developed by the ETSI(European Telecommunications Standards Institute). Currently, TETRA was set Digital TRS in electric power If wireless backbone network. In this time, we use many company's TETRA modem. So, TETRA modem performance evaluation is very important. TETRA modem use two type of Data transfer mode. One is Packet Data using UDP/IP. and the other is SDS(Short Data Service). In this paper, We generate Packet Data using Traffic Generator module. Packet Data transfer 1000 times each 10 bytes to 400 bytes. We analyze transmission delay time, success rate and standard deviation.

• Journal of Internet Computing and Services
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• v.9 no.4
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• pp.1-10
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• 2008

In this paper, a packet scheduling algorithm, called the modified PLFS, is proposed for real-time traffic in IEEE 802.22 WRAN systems. The modified PLFS(Packet Loss Fair Scheduling) algorithm utilizes not only the delay of the Head of Line(HOL) packets in buffer of each user but also the amount of expected loss packets in the next-next frame when a service will not be given in the next frame. The performances of the modified PLFS are compared with those of PLFS and M-LWDF in terms of the average packet loss rate and throughput. The simulation results show that the proposed scheduling algorithm performs much better than the PLFS and M-LWDF algorithms.

• KSII Transactions on Internet and Information Systems (TIIS)
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• v.12 no.2
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• pp.579-603
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• 2018

In this paper we propose a QoS-enhanced intelligent stochastic optimal fair real-time packet scheduler, QUEST, for 4G LTE traffic in routers. The objective of this research is to maximize the system QoS subject to the constraint that the processor utilization is kept nearly at 100 percent. The QUEST has following unique advantages. First, it solves the challenging problem of starvation for low priority process - buffered streaming video and TCP based; second, it solves the major bottleneck of the scheduler Earliest Deadline First's failure at heavy loads. Finally, QUEST offers the benefit of arbitrarily pre-programming the process utilization ratio.Three classes of multimedia 4G LTE QCI traffic, conversational voice, live streaming video, buffered streaming video and TCP based applications have been considered. We analyse two most important QoS metrics, packet loss rate (PLR) and mean waiting time. All claims are supported by discrete event and Monte Carlo simulations. The simulation results show that the QUEST scheduler outperforms current state-of-the-art benchmark schedulers. The proposed scheduler offers 37 percent improvement in PLR and 23 percent improvement in mean waiting time over the best competing current scheduler Accuracy-aware EDF.

• Journal of the Korea Society of Computer and Information
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• v.15 no.1
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• pp.129-137
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• 2010

In this paper, we propose a packet prioritization scheme using preamble signal and backoff time which is designed to provide a differentiated channel occupation with a packet priority in wireless sensor networks. This scheme aims at enabling QoS such as fast delivery of high priority packets by reducing their backoff time as well as thus securing higher channel occupation. We expect that it could also improve the channel utilization of the entire network by avoiding unnecessary channel contention. For the purpose, we add new features of multiple queue and preamble modification to the TinyOS based B-MAC. This scheme achieves 82-88% reduction in delivery time of high priority packets, thus it enables realtime support for urgent applications.